path: root/configs
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authordvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b>2011-02-22 23:04:49 +0000
committerdvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b>2011-02-22 23:04:49 +0000
commitf27e928f0588f5cbf85ac8202cef912efcc51a9c (patch)
treeb061487de973558358757bd1b6e457aaccf41638 /configs
parent70442b4e1767b35ed1699d27cfc24109c617f445 (diff)
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308582 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs')
1 files changed, 71 insertions, 0 deletions
diff --git a/configs/codecs.conf.sample b/configs/codecs.conf.sample
index c8caeab..4404d4a 100644
--- a/configs/codecs.conf.sample
+++ b/configs/codecs.conf.sample
@@ -63,3 +63,74 @@ pp_dereverb_level => 0.3
; this determines whether to perform generic PLC
; there is a minor performance penalty for this
genericplc => true
+; Generate custom formats for formats requiring attributes.
+; After defining the custom format, the name used in defining
+; the format can be used throughout Asterisk in the format 'allow'
+; and 'disallow' options.
+; Example: silk8 is a predefined custom format in this config file.
+; Once this config file is loaded, silk8 can be used anywhere a
+; peer's codec capabilities are defined.
+; In sip.conf 'silk8' can be defined as a capability for a peer.
+; [peer1]
+; type=peer
+; host=dynamic
+; disallow=all
+; allow=silk8 ;custom codec defined in codecs.conf
+; Custom formats can only be defined at startup. Any changes to this
+; file made after startup will not take into effect until after Asterisk
+; is restarted.
+; Default Custom SILK format definitions, only one custom SILK format per
+; sample rate is allowed.
+fec=true ; turn on or off encoding with forward error correction.
+ ; On recommended, off by default.
+packetloss_percentage=10 ; Estimated packet loss percentage in uplink direction. This
+ ; affects how much redundancy is built in when using fec.
+ ; The higher the percentage, the larger amount of bandwidth is
+ ; used. Default is 0%, 10% is recommended when fec is in use.
+maxbitrate=10000 ; Use the table below to make sure a useful bitrate is choosen
+ ; for maxbitrate. If not set or value is not within the bounds
+ ; of the encoder, a default value is chosen.
+ ;
+ ; sample rate | bitrate range
+ ; 8khz | 5000 - 20000 bps
+ ; 12khz | 7000 - 25000 bps
+ ; 16khz | 8000 - 30000 bps
+ ; 24khz | 20000- 40000 bps
+ ;
+;dtx=true ; Encode using discontinuous transmission mode or not. Turning this
+ ; on will save bandwidth during periods of silence at the cost of
+ ; increased computational complexity. Off by default.